Hi,
Is anybody there who can give me information which things are supported in
Asterisk on Dialogic E1-Cards?
Does anybody use a Dialogic card in Asterisk? (Not the DIVA, i know eicon
is using some ISDN-Stack which is working in Asterisk)
I mean the really old Dialogic cards.
thanks
Nico
Try this:
sox -r 44100 -w -s -c 1 myfile.raw -r 8000 -c 1 myfile.wav
I hope this helps
l.
On Tue, 13 Nov 2007 15:26:07 +0100, Gary [EMAIL PROTECTED] wrote:
I used ChanSpy( ) recorded some test conversations. It has .raw
extension.
What kind of audio file is this? How can I play it?
dear
I am searching for the company like pipemedia
(legend.co.uk) in USA, or other european countris.
I tested, didex.org, but pipemedia is more advance
tele-communication company.
please tell me, if you know.
thanks
best
Mani
I downloaded packages for 1.4.
Shoudl I now install asterisk, zaptel, libs for unicall as it is typed at
http://www.voip-info.org/tiki-index.php?page=Asterisk+MFC+R2 ?
astunicall package already include zaptel, asterisk and all the Steve
underwood libraries. However, personally, since Elastix
14 nov 2007 kl. 01.14 skrev Marco Mouta:
${DIALSTATUS} will be one of:
You can also use the SIPPEER() dialplan function to check the status
of the peer,
provided you have a qualify= setting for that peer to enable health
and latency
checking. If the peer is unavailable, you can do your
This is propably because you've registered in sip.conf without an
extension in the register= configuration
line. Check the syntax.
If you can't fix that, use the SIP_HEADER() dialplan funcion to get
the value of the To: header and
use that as an extension.
/Olle
---
* Olle E Johansson -
Hi,
Tomorrow, Friday Nov 16th, 2007 at 12:30 PM, we'll be exploring a
simple, well-commented example of an AGI script for asterisk. I have
absolutely nothing against GUI, but if you want to unleash the real
power of asterisk, you'll need to get into AGI (or pay someone else to
do it). Because
The http://voipUsersConference.com/ning seems not to be working.
l.
On Thu, 15 Nov 2007 12:03:56 +0100, randulo [EMAIL PROTECTED]
wrote:
Hi,
Tomorrow, Friday Nov 16th, 2007 at 12:30 PM, we'll be exploring a
simple, well-commented example of an AGI script for asterisk. I have
absolutely
Hi
Here is my setup:
USER -- PSTN - Asterisk A IAX2 Trunk Asterisk
B - SER Asterisk C
I'm not able to receive DTMF passed by USER on Asterisk C.
All my asterisk boxs are configured with same DTMF type (auto) but no luck.
Please help on this issue.
On Nov 15, 2007 12:55 PM, Lenz [EMAIL PROTECTED] wrote:
The http://voipUsersConference.com/ning seems not to be working.
/me SLAPS his head HARD
http://voipUsersConference.org/ning
repeats to self .org .org .org .org
It's actually a redirect to this: http://food4wine.ning.com
Sorry! At least
Hi，
Could you capture the the UDP package in all of your server, Asterisk A,
Asterisk B, ser, Asterisk C.
And you can find that server who lost the DTMF (RTP EVENT).
Amy
2007-11-15
发件人： Arun Kumar
发送时间： 2007-11-15 20:30:45
收件人： Asterisk Users Mailing List - Non-Commercial Discussion; SER
i am have installed TE120P e1/t1 card but i have never install TE220P PCI
express card so i just want to know it will install same mathod like TE120P
card or some different type of installtion for this TE220 card just i wann know
about configuration option or zaptel compilation option for TE220
All,
I have noticed that placing a call in the outgoing spool during a reload
the call may fail. Try the call again after the reload is done and it will
complete.
This seems like a bug. During a reload calls should be suspended or
something?
Thoughts?
Jerry
Dear all
anybody have idea of this 2 card and performance vise which one
is suggestable ???
If you had done a little bit of legwork, you'd have noticed that the
TE220 has a PCI-Express interface while the TE210 has a 3.3V PCI
interface. There is no difference between them in terms
Anybody know how to get the tmobile blackberry 8320 to connect to asterisk
via voip? The phone is wifi enabled and does use voip for tmobile.
TIA,
Jon
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asterisk-users mailing
If you want peep every 15s, you should do:
[some_context]
exten = _X.,1,Set(LIMIT_WARNING_FILE=beep)
exten =
_X.,2,Dial(Local/mixmoncontext/#{EXTEN}||L(3600:3600:15000)\n)
That is it... very much appreciated.
Tony Plack
___
--Bandwidth
On Thu, 15 Nov 2007, Benjamin Jacob wrote:
well.. if nothings working.. try putting in debug lines urself in the
code.. say
use system calls to write some debugging data into some temporary file
in ur perl code.
I'm a big fan of
syslog(LOG_ERR, I expected %d, but I got %d, foo,
[EMAIL PROTECTED] wrote:
Hi,
Is anybody there who can give me information which things are supported in
Asterisk on Dialogic E1-Cards?
Does anybody use a Dialogic card in Asterisk? (Not the DIVA, i know eicon
is using some ISDN-Stack which is working in Asterisk)
I mean the really old
On Thu, 2007-11-15 at 05:34 +0100, Patrick wrote:
On Wed, 2007-11-14 at 09:06 -0500, Anciso, Roy wrote:
The Cisco Documentation states that you can modify standard and
nonstandard softkey templates. They may not be xml files. I just
assumed they were xml since that is what is used to
Mohammad Shokuie wrote:
Hi Erik,
By firefox i mean a Hotmail web mail, it means there is no mail client. I
dont know if there would be any difference if i subscribe and use other
mails like gmail!
Regards.
--
M. Shokuie Nia
Hotmail web mail is an email client, just not a local
At 06:42 11/15/2007, =?gb2312?B?xL7Evg==?= wrote:
Hi£¬
Could you capture the the UDP package
How is this done?
in all of your server, Asterisk A, Asterisk B, ser, Asterisk C.
And you can find that server who lost the DTMF (RTP EVENT).
--
Amy
2007-11-15
--
·¢¼þÈË£º Arun
Ok, probably a dumb question. I believe I already I know the answer, but
thought I would get feedback from others.
One of the issues with user devices at the end Asterisk is dialing time
out. This is a parameter within each hardware device. So if I set it to 3
seconds it appears from the
Hi, All
Here is the install script that I promised. Let me know if there are any
problems. I did not include nv_faxdetect in this yet, I am still working on
that. The hylafax-iaxmodem script I posted earlier should work on CentOS 5.
You can always press # at the end of your number to send it to Asterisk.
Mindaugas Kezys
http://www.kolmisoft.com
MOR - Advanced Billing for Asterisk PBX
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jim Houser
Sent: Thursday, November 15, 2007 6:38
Correction to this post. Here is the script for CentOS 5.
http://jonnt.users.taylortelephone.com/asterisk/iax-hylafax-setup-centos5.sh
Jonn
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jonn R Taylor
Sent: Thursday, November 15, 2007 10:49 AM
To:
Last message received at
2007-11-14 18:02:04 GMT
Regards,
Philipp Kempgen
--
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Let's use IT to solve problems and not to create new ones.
Asterisk? - http://www.das-asterisk-buch.de
Geschäftsführer: Stefan
On Debian the Asterisk Makefile does
/usr/sbin/update-rc.d asterisk start 10 2 3 4 5 . stop 91 2 3 4 5 .;
which results in a /etc/rc2.d/S10asterisk being written.
I think S10 is too early.
bind9 : S15
mysql : S19
zaptel: S20
ntp : S23
What bothers me most is that mysql is not up when
which panasonic system is this? as I have succesfully integrated
panasonic systems in the past without a
problem
On 11/14/07, jorain [EMAIL PROTECTED] wrote:
Hi all,
I have an existing panasonic analog pbx in use and a asterisk server with
digium tdm400p(2 fxs and 2 fxo).
channel 1 - fxs -
Good morning,
we have two configured queues on our asterisk server in the company
and have the following problem with the caller positions in the queue.
In our test case we have 5 agents logged in the queue and five test
persons calling the queues. Sometimes, this case unfortunately cannot
more
Does anyone store gsm files on a shared server so multiple asterisk boxes
can access the common gsm files?
I want to do this so they can be updated easily, but wanted to make sure I
wouldn't run in to any unforeseen problems. If anyone has done this could
you tell me what you used and if you had
We've got an SPA-2100 connected to * and then into a paging system on
one of the FXS ports. We are having an issue where the paging system
doesn't hang up the line, so it stays offhook forever and obviously
makes in unusable. The paging company says that the SPA needs to hangup
the line once
If you want peep every 15s, you should do:
[some_context]
exten = _X.,1,Set(LIMIT_WARNING_FILE=beep)
exten =
_X.,2,Dial(Local/mixmoncontext/#{EXTEN}||L(3600:3600:15000)\n)
[mixmoncontext]
exten = _X.,1,MixMonitor...
In [some_context] use L option variables:
*
On Thu, 15 Nov 2007, Jay R. Ashworth wrote:
On Thu, Nov 15, 2007 at 07:29:29PM +0100, Olivier wrote:
Which is the best way to manage logs ?
Would you centralize and mix logs from Linux, Asterisk, syslog and
others
or keep them separate ?
In my experience, it's easier to combine
On Thursday 15 November 2007 11:33:46 Ryan M. Colbert wrote:
Periodically, maybe once or twice every few weeks, we see our instance of
Asterisk 1.4.7 just close out without warning and we have to reload the
module. We're running CentOS. Has anyone else seen this before?
Core show version:
Hi,
Which is the best way to manage logs ?
Would you centralize and mix logs from Linux, Asterisk, syslog and others
or keep them separate ?
Cheers
___
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asterisk-users mailing list
To
I have not been able to get two B-channel transfer to work on DMS100 PRI. I
consistently get the following errors:
[Nov 6 11:12:49] ERROR[2774]: chan_zap.c:8178 zt_pri_error: ROSE RETURN
ERROR:
[Nov 6 11:12:49] ERROR[2774]: chan_zap.c:8178 zt_pri_error:OPERATION:
RLT_OPERATION_IND
[Nov 6
On Thu, Nov 15, 2007 at 07:29:29PM +0100, Olivier wrote:
Which is the best way to manage logs ?
Would you centralize and mix logs from Linux, Asterisk, syslog and others
or keep them separate ?
In my experience, it's easier to combine them all into one syslog
server, and then utilize
I posted to the list earlier this week about this very issue. This
reinforces my thought that it is a bug in 1.4.7.
Since upgrading the box to 1.4.13 the issue resolved itself.
I have not opened a issue in the tracker as I hadn¹t had time to try and
replicate the issue.
On 16/11/07 5:32 AM,
What do you mean by reload. Please be specific. Reload or restart.
Reload which module? or did you mean restart Asterisk?
On Nov 15, 2007 7:12 AM, Jerry Geis [EMAIL PROTECTED] wrote:
All,
I have noticed that placing a call in the outgoing spool during a reload
the call may fail. Try the call
Pepo wrote:
Hi friends.
How do I can use Asterisk 1.4 with LDAP? I need it because the system must
use
just one password for each user for everything.
A lot of thanks.
What exactly in asterisk would your LDAP be authenticating? Sip
registrations? Thats a device, not a user.
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